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Asterisk Audio and Video Capabilities - Asterisk Project ...
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Audio+and+Video+Capabilities
Asterisk supports a variety of audio and video media. Asterisk provides CODEC modules to facilitate encoding and decoding of audio streams. Additionally file format modules are provided to handle writing to and reading from the file-system. The tables on this page describe what capabilities Asterisk supports and specific details for each format.
Recording Audio for Asterisk | Redial
http://www.itp-redial.com/class/weekly-notes/week2-notes/recording-audio-for-asterisk
Recording Audio for Asterisk Standards The audio standard for most Voice over IP is 8000Hz, 16Bit, Mono. Asterisk is going to expect that all audio conforms to this standard. There are a lot of codecs that are used to compress the audio, the most common being ULAW, GSM, and G729. The PSTN will only accept ULAW or G729.
How To Convert Audio Files for Use With Asterisk
http://www.verterion.com/asterisk/convert-audio-files-use-asterisk/
For Asterisk-based systems, this depends on the CODEC you are using. 90% of the time, the g711 uLAW codec is used, which means the sound file must be …
12.2 jail, where is Asterisk audio? | TrueNAS Community
https://www.truenas.com/community/threads/12-2-jail-where-is-asterisk-audio.98788/
#1 I am testing an Asterisk jail. I install asterisk18 (because LTS). Phones working good. Only problem i have, no asterisk audio. The playback functionality. Just to be sure i tried another jail with asterisk 16. Same results. Is this because of no audio support on TrueNAS?
Asterisk webrtc no audio - Asterisk WebRTC - Asterisk ...
https://community.asterisk.org/t/asterisk-webrtc-no-audio/91333
I config asterisk for webrtc and sipjs.com. so that work for SimpleUser configuration. with SimpleUser every thing is Ok. but when I use UserAgent , connection has established without audio .this is asterisk console when User3 call User2 :
No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
2) Not answered calls: in your dialplan ( /etc/asterisk/extensions.conf) you need to answer calls before they can send any audio (RTP), make sure every call runs through an Answer ().
Correcting One-way Audio with a VoIP Call - Asterisk
https://www.asteriskpbxsystems.com/troubleshoot-oneway-audio.html
Connect the ATA/IAD or other SIP device directly to the first device on the LAN such as the modem. Check for normal operation and good two-way audio. If you experience one-way audio or do not receive dial tone, then the issue will most likely be SIP unfriendly NAT, or a firewall present that is not allowing the correct VoIP packets to cross.
Guru Audio Convert - Asterisk Guru
https://www.asteriskguru.com/tools/audio_conversion.php
Online audio conversion tool - Guru Audio Convert : wav import files are currently not working, we are working on it. Input File * ... Asterisk; Asterisk 1.4.20 Released section: Asterisk; Asterisk 1.4.20-rc2 Released section: Asterisk; Asterisk 1.4.20-rc1 Now Available
AudioSocket - Asterisk Project - Asterisk Project Wiki
https://wiki.asterisk.org/wiki/display/AST/AudioSocket
The singular design goal of AudioSocket is to present the simplest possible audio streaming protocol, initially based on the constraints of Asterisk audio. Each packet contains a three-byte header and a variable payload. The header is composed of a one-byte type and a two-byte length indicator.
Asterisk 15: Multi-stream Media and SFU ⋆ Asterisk
https://www.asterisk.org/asterisk-15-multi-stream-media-sfu/
Now that Asterisk has been setup, start it, and then open the “index.html” file (in either Firefox or a Chrome/Chromium browser) that’s in the top level directory of CMP2K. You should see something like the following: It’s quite a fancy application I know!
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